Aeri, WebRTC reliability cloud
Drop one line of SDK into any peer connection. Get a full session timeline, plain-English root causes when a call fails, and preflight checks before you ship. Vendor neutral across Cloudflare, LiveKit, Daily, Twilio, Agora and coturn. The browser lab needs no account.
SIP Ladder ->
Paste a raw SIP or Asterisk log, get a colour-coded ladder with a plain-English diagnosis for codec mismatches, auth loops and NAT. Runs fully in your browser.
Flagship sandboxSIP Lab ->
Build a call visually. Drag phones, PBXs, trunks and carriers onto a canvas, wire them up, set codecs and auth, then run it and watch it connect or fail.
Live referenceSIP code decoder ->
Paste any SIP response code for a plain-English explanation, common causes and fixes.
Live referenceSIP code reference ->
Every SIP response code from 100 to 608 as its own page, with who sends it, the real causes, and how to fix it. The full library behind the decoder.
Live test lineDTMF test line ->
Call our free number to hear your caller ID, STIR/SHAKEN attestation level, and test your DTMF keypad tones.
ICE candidate inspector ->
See your WebRTC ICE candidates live, discover your NAT situation, and test your own STUN or TURN server from the browser.
Live SDPSDP analyzer ->
Paste a WebRTC SDP offer or answer for a plain-English breakdown of codecs, ICE, DTLS and media directions.
Live qualityConnection quality analyzer ->
Run a live connection and watch bitrate, round trip time, jitter, packet loss and MOS graph in real time. Add TURN to measure your real network.
Live devicesMedia device tester ->
Test your camera, meter your microphone, play a stereo speaker tone, and list every media device. Nothing recorded or uploaded.
Number lookup ->
Check carrier, line type, fraud score, portability and country for any phone number instantly.
Live reputationSpam score check ->
Check if a number is a scammer using pattern matching, fraud scoring and community reports.
Live formatE.164 converter ->
Convert any number between E.164, national, international and RFC 3966 formats. Works worldwide.
Scam pattern checker
Match a number against known scam patterns like Macau scam, LHDN and Bank Negara impersonation.
Caller ID reputation
Full spam reputation report covering history, carrier hops and community reports.
Community scam board
Submit and search scam numbers reported by the community. Crowdsourced protection.
Bulk number lookup
Upload a CSV of numbers and get carrier and line type data for all of them at once.
Best time to call
Enter a number to get the caller's local timezone and whether it is a reasonable hour.
Country code lookup
Search any country for its dialling prefix, number format rules and emergency numbers.
Mock API ->
A free JSON API with predictable fictional cases, customers and orders, plus deliberate 404, 500 and slow endpoints so you can prove your error handling works.
Live convertIVR audio converter ->
Convert any MP3 or WAV to the exact format your PBX needs. One-click presets for Asterisk, FreePBX, 3CX and Cisco CUCM. Runs in your browser.
Broken UAS endpoints
SIP endpoints that misbehave on demand: 407 loops, 503 with Retry-After, late SDP, one way audio. Test your error handling against the ugly cases.
VoIP bandwidth calc
Choose a codec and concurrent call count to get the exact bandwidth needed in Kbps.
MOS score calculator
Enter jitter, latency and packet loss to get an estimated call quality MOS score.
Learn Telephony & WebRTC
A growing, structured course on how phone calls, SMS and real-time communication actually work. Plain-English guides with interactive diagrams, organised beginner to advanced. Start anywhere, follow the path.
Enter the course, 30+ guides ->What is WebRTC ->
The technology behind every browser video call, explained in plain English. No app, no plugin, and it is probably already running in your browser.
BasicsWhat is SIP ->
The language phones use to start, manage and end a call. Readable text messages you can actually follow, with a real example.
SecurityHow call spoofing works ->
Why scammers can fake any caller ID, covering the SS7 vulnerability and how to protect yourself.
10 stepsSTUN vs TURN ->
Why most calls connect directly, why some cannot, and what a TURN server actually relays.
14 stepsHow a SIP call works ->
INVITE, 100 Trying, 180 Ringing, 200 OK, ACK, RTP and BYE, all with real SIP messages.
BasicsCircuit vs packet switching ->
Why a WhatsApp call is free and a normal mobile call is not. The two ideas every modern call is built on.